1. Field of the Invention
This invention relates to a method and apparatus for processing telephone calls in a packet-based telephony service. In particular, but not exclusively, the invention relates to the processing of such calls in a multi-carrier telecommunications network which includes a plurality of independent network operators each providing telephony services to subscribers via their carrier networks.
2. Description of the Related Technology
Conventional multi-carrier telecommunication networks typically have an Incumbent Local Exchange Carrier (ILEC) which owns most of the local telephone infrastructure and other local carriers, known as Competitive Local Exchange Carriers (CLECs) which compete with the ILEC. A CLEC provides its services by subcontracting network facilities from the ILEC and provides alternative services to subscribers.
The CLEC may provide a packet-based service over a connection such as a digital subscriber line (DSL) connection. A DSL connection is suitable for transmitting analogue voice calls and packet-based data simultaneously down the same telephone line. Voice calls can also be sent as packet-based data using the Internet Protocol (IP), and such calls are known as Voice over Internet Protocol (VoIP) calls.
The International Telecommunications Union standards for telecommunications (ITU-T) body specifies the Signaling System #7 (SS7) protocol for digital exchanges with integrated services. The SS7 protocol provides international data network and signaling protocols that control calls passing through a PSTN. SS7 employs out-of band signaling to transmit messages between switches and other network entities via circuit-switched connections such as redundant data links. SS7 includes a sub-protocol known as the Integrated Services Digital Network User Part (ISUP).
The ISUP protocol is a functional part of SS7 which defines the procedures used for transfer of call setup and teardown signaling information between signaling points over a PSTN. ISUP is used for both ISDN and non-ISDN calls. Different variants of ISUP exist, for example the European Telecom Standardization Institute (ETSI) ISUP variant is used in Europe.
Each ISUP variant specifies a mandatory part for fixed-length mandatory parameters and an optional part for variable length optional parameters. An ISUP Initial address Message (IAM) is sent in the forward direction by each switch in the path from calling party to called party. An IAM contains the called party number in the mandatory variable part and may contain the calling party name and number in the optional part.
In multi-carrier telecommunications networks, signaling information must be passed between the different carriers. Various protocols have been developed by the Internet Engineering Task Force (IETF). The IETF's Signaling Transport working group has developed the Session Initiation Protocol (SIP) which can be employed to address the transport of packet-based signaling for VoIP calls.
SS7 messages can be either encapsulated in their entirety within a SIP message body for transparency of ISUP signaling, or the ISUP information can be translated into a SIP message header for the routing of SIP messages by entities such as proxy servers which do not understand the ISUP protocol. The ISUP IAM information can be incorporated into SIP call setup requests known as Invites. The reverse process, converting SIP messages to ISUP messages can be carried out at an IP-to-PSTN interface. In this way, the legacy SS7 signaling information can be preserved for telephone calls made between a mixture of PSTN and IP endpoints.
United States patent application US2004/0192292, describes apparatus for selectively connecting an analogue telephone circuit to either Public Switched Telephone Network (PSTN) based telephony services or packet-based telephony services. The user is able to selectively place or receive calls via either type of service and routing and billing facilities between the carriers must be configured accordingly. The user can receive calls either via a telephone number corresponding to a PSTN based service or via a different telephone number corresponding to the Internet based service. When placing a call, the user can choose which type of service they would like the call to be routed through, each service having a different connection procedure. If a calling party identification service such as Calling Line Identifier (CLI) is employed, the calling party will be identified at the call destination point either by a PSTN based service number or by the packet-based service number. The process of routing telephone calls for multi-service users in a multi-carrier telecommunications network according to the prior art is now described with reference to FIG. 1.
A multi-service user accesses telephone services via a telephone 40 which has access to a multi-carrier network, including a circuit-switched carrier network and a packet-based carrier network. The circuit-switched carrier network 44 is typically a PSTN and the packet-based carrier network 70 provides a VoIP service as shown in FIG. 1 by items 44 and 70 respectively. A Softswitch 42 in the packet-based carrier network interfaces with the PSTN. Softswitches are entities or clusters of entities, also known as Media Gateway Controllers (MGCs) and call agents. Softswitches provide the intelligence that controls packet-based telephony services, including the ability to select processes that can be applied to a call, routing for a call within the network based on signaling and subscriber database information, the ability to transfer control of the call to another network element and management functions such as provisioning, fault detection and billing. Softswitches also provide the architecture for enabling conversion between both signaling protocols such as SS7 and SIP and circuit-switched and packet-based voice calls. For ease of explanation, it is hereafter assumed that a Softswitch is one network entity, although in practice this may be a distributed set of entities.
A media gateway is responsible for handling the media data for calls, the media data being the data packets which contain the payload of the call (e.g. voice data) as opposed to the signaling data packets used for controlling the call. The media gateway typically includes communications switch equipment and operates between a packet-based part of the telecommunications network and the Public Switched Telephone Network.
A multi-service user subscribes to two or more telephony services. Each can provide their own telephony party identifier and the user thus has two or more different telephony party identifiers (TPIs), one supplied by each carrier network operator. These TPIs would typically be telephone dialing numbers allocated to the user by each network operator, and which are used by that network operator's carrier network to route telephone calls to the user.
When a call is made to the multi-service user via the circuit-switched carrier network 44 on the telephone dialing number allocated by the circuit-switched carrier network, a first TPI (TPI1) is placed in a called party number field in the signaling information 58 for that call according to the SS7 (or equivalent) signaling protocol as outlined above. The signaling information 60 containing TPI1 is then used by the circuit-switched carrier network 44 signaling infrastructure to route the call for the user via the circuit-switched carrier network, without passing through the packet-based carrier network 70.
If a call is made to the user via the circuit-switched carrier network 44 on the telephone dialing number allocated by the packet-based carrier network 70, a second TPI (TPI2) is placed in a called party number field in the signaling information 62 for that call according to the SS7 (or equivalent) signaling protocol as outlined above. The signaling information containing TPI2 is then passed as signaling information 64 to the Softswitch 42 responsible for processing calls for the packet-based carrier network. The Softswitch 42 then recognizes from TPI2 that the call is for the telephone 40 of the user and passes on the signaling information containing TPI2 as outgoing signaling information 66, which is sent to the user using the SIP protocol.
When a call is made to the multi-service user via the packet-based carrier network 70 on the telephone dialing number allocated by the packet-based carrier network, TPI2 is placed in the signaling information 68 for that call according to the SIP (or equivalent) signaling protocol as outlined above. The Softswitch 42 receives incoming signaling information 68 containing TPI2 which is then passed as outgoing signaling information 66 on to the telephone 40 of the user. Both the incoming signaling information 68 and the outgoing signaling information 66 are sent using the SIP protocol.
If a call is made to the multi-service user via the packet-based carrier network 70 on the telephone dialing number allocated by the circuit-switched carrier network, TPI1 is placed in the signaling information 72 for that call according to the SS7 (or equivalent) signaling protocol as outlined above. The Softswitch 42 receives the incoming signaling information 72 containing TPI1 and recognizes that the call should be routed to the external network carrier. The incoming signaling information 72 in the SIP protocol messaging is converted into the SS7 protocol and is then passed as outgoing signaling information 74 from the Softswitch 42 to be processed by the circuit-switched carrier network 44.
When a call is made by the multi-service user via the circuit-switched carrier network 44, TPI1 is placed in the signaling information 46 for that call according to the SS7 (or equivalent) signaling protocol as outlined above. The signaling information 48 containing TPI1 is then passed on to the telephone 76 of the destination party via the circuit-switched carrier network 44.
When a call is made by the multi-service user via the packet-based carrier network 70, TPI2 is placed in the signaling information 50 for that call according to the procedure outlined for the SIP (or equivalent) signaling protocol above. The Softswitch 42 receives the incoming signaling information containing TPI2. The Softswitch will judge whether the call should be routed via the circuit-switched carrier network or via the packet-based carrier network. This judgment may depend on whether the destination party is also a subscriber to the packet-based carrier network. If the destination party is not a subscriber to the packet-based carrier network, then the call will be routed via the circuit-switched carrier network, whereas if the destination party does subscribe to the packet-based carrier network, then the call may be routed via the packet-based carrier network. If the call is to be routed via the circuit-switched carrier network, outgoing signaling information 54 containing TPI2 is passed from the Softswitch 42 on to the circuit-switched carrier network 44 as SS7 protocol messaging and is then passed as signaling information 56 on to the telephone 76 of the destination party via the circuit-switched carrier network 44.
If the call is to be routed via the packet-based carrier network, outgoing signaling information 52 containing TPI2 is passed from the Softswitch 42 via the packet-based carrier network 70 to the telephone 78 of the destination party.
As will be appreciated from the above, a problem with conventional multi-service telecommunications networks is that each carrier supplies the subscriber with a different telephone number for their particular service. The routing of data and/or voice calls to and from individual subscribers then involves a plurality of telephone numbers, which can be highly inconvenient and confusing to other users of the system.
One solution would be to move over entirely to a packet-based service and use the telephony party identifier provided by the packet-based service exclusively. However, most users would wish to maintain an alternate telephony service in case the packet-based service is for some reason not available for a period of time, in particular when there is an emergency. It would be desirable to provide an improved method and apparatus for processing telephone calls for users having access to a plurality of telephony services.